CUBE Licensing in 29XX/39XX Routers

Is Unified Border Element licensing enforced on the 2900 and 3900 ISRs? If so, what command can I use to determine how many sessions my VGs are licensed for? A show version just says I have the UC license.

1. Configure CUBE for media flow through. In this Mode CUBE acts as a true B2BUA. Advantages you get include address hiding and security becaue CUBE terminates and re-originate both signalling and Media. In this mode CUBE becomes a point of demarcation from th external world.

2. Configure CUCM to use Delayed Offer and CUBE to convert delayed offer to ealry offer...This prevents the need for you to use MTP to send Early offer on CUCM

voice service voip

early-offer forced

3. Configure DTMF signalling method on sip trunk to "No preference" This setting allows Unified CM to make an optimal decision for DTMF and to minimize MTP allocation.

4. Configure your CUBE to meet the requirements of your ITSP. Ask if they have configuration templates or any specific configuration they like you to use. This will save you time troubleshooting. Most of them dont use the default port 5060 because of security, confirm with your proivider what ports they use.

voice service voip

allow-connections sip to sip

sip

early-offer forced

header-passing

error-passthru

5. Use SIP to SIP...Use end to end sip. CUCM---sip---CUBE--sip----ITSP

6. Create a Trusted list of IP addresses on your CUBE is your CUBE IOS is 15.1 .2(T) and above.

voice service voip

ip address trusted list

ipv4 203.0.113.100 255.255.255.255

ipv4 192.0.2.0 255.255.255.0

This is imprtant because sometimes your ITSP will send you a single ip address for signalling and will then send media on a different IP adress. So get all the IP address your ITSP is using and add them to the trust list as shown above

You may need to configure sip normnalization to modify sip headers, CLI etc. A good example is during call forwarding. You may need to change your diversion headers to match the CLI your provider is expecting.. if your call forwarding is failing during testing this may be the reason..We can help you with this.

9. Media Resources

Plan your solution properly. Consider if you will need Xcoders, MTP, Conference bridge etc. You may avoid the need for xcoders if you confure your regions properly and use voice class codecs on your sip profiles. It is important to know if there are any endpoints in your network that do not support dtmf relay rtp-nte. You can avoid the use of MTP if you configure your dial-peers to have multiple dtmf types for thos phones that do not support rtp-nte

e.g

dial-peer voice 1 voip

session protocol sipv2

dtmf-relay rtp-nte digit-drop sip-kpml (if your phones support kpml..then this will be used)

If in your environment you will need to do xcoding or CFB then ensure you have PVDMS

.10.FAX

If you have FAX in your network, determine what fax protocol your sip provider supports. Dont assume. Ask them and confirm in writing what they support. I have seen legal cases because of fax failures over sip trunks

Configure your FAX devices in seperate device pools and use porefix to route calls using G711 only. Even if you are using T38, ensure your fax use G711 to establish the voice calls